Issues 21000 - 21999

[..]
ASTERISK-21002: Originate without Exten header does not work
ASTERISK-21003: MOH keeps playing for the fist participant, if two participants connect at the same time to an empty conference
ASTERISK-21004: Open Blockers for 1.8.21.0
ASTERISK-21005: Open Blockers for 11.3.0
ASTERISK-21006: unsupported host os "linux-gnueabihf"
ASTERISK-21007: Remove Non Real time SIP Peers Automatically
ASTERISK-21009: xmpp_pubsub_unsubscribe: Could not create IQ when creating pubsub unsubscription on client
ASTERISK-21011: Asterisk to Asterisk IAX2 trunk registration does not register
ASTERISK-21012: Memory Leak on res_calendar (icalendar)
ASTERISK-21013: Security Vulnerability: sip username disclosure
ASTERISK-21014: logger.c Call_ID 'bound' or 'removed' DEBUG messages spammed during a feature code attended transfer
ASTERISK-21016: One-way audio with JABBER correlates with 'res_xmpp.c: JABBER: socket read error' in debug
ASTERISK-21019: PBX Cant Initialize (device_state_db - line 10492)
ASTERISK-21020: Add support for legacy sip.conf configuration in res_sip's supported sorcery backends
ASTERISK-21021: SQL script to create queue_log table in PostgreSQL
ASTERISK-21024: Implement stasis-http GET /api/channels
ASTERISK-21025: Implement stasis-http GET /api/channels/{channelId}
ASTERISK-21026: Implement stasis-http POST /api/channels/{channelId}/answer
ASTERISK-21027: Implement stasis-http DELETE /api/channels/{channelId}
ASTERISK-21028: Implement stasis-http POST /api/channels/{channelId}/continue
ASTERISK-21035: [patch] - features.conf in static realtime requires distinct cat_metric for each parking lot
ASTERISK-21036: Jitter Buffer log file creation doesn't account for multiple slashes in DAHDI channel names
ASTERISK-21037: skinny global vmexten and immed dial dont reset on module reload
ASTERISK-21038: CLI: "core set debug channel" auto-complete returns "all", but not the names of available channels
ASTERISK-21039: ODBC functions time out
ASTERISK-21040: Deadlock involving chan_sip.c, pbx.c and autoservice.c, locking on chan and &conclock
ASTERISK-21041: Asterisk crashes during a frame copy while receiving a fax
ASTERISK-21042: [patch] - pbx_spool: callfile variables overriding/lost in __ast_request_and_dial()
ASTERISK-21043: Motif/XMPP/Google Voice based calls keep ringing after re-establishing XMPP connection after socket error
ASTERISK-21044: Call quality degrades after thousands of calls over a short period
ASTERISK-21045: Session refresh reinvites an in progress T.38 dialog back to G.711
ASTERISK-21046: res_xmpp refcount issue
ASTERISK-21047: asterisk crashes during an attended transfer SIP -> SIP -> DAHDI
ASTERISK-21050: Asterisk crash during startup - issues with dlclose() return code checks and module loading registration
ASTERISK-21051: Bridge API Enhancements: Refactor callers of ast_bridge_call to use Bridging API model
ASTERISK-21052: Bridge API Enhancements: Implement threading model for bridge management thread
ASTERISK-21054: Bridge API Enhancements: Add roles to the bridging model
ASTERISK-21057: Bridge API Enhancements - add Stasis-Core events
ASTERISK-21058: Bridge API Enhancements - rework Local channels/Local channel bridging
ASTERISK-21059: Bridge API Enhancements - Refactor the Park family of applications
ASTERISK-21061: Nortel I2004 unwanted autoanswer
ASTERISK-21062: Pedantic should also be per extension directive
ASTERISK-21063: Fix some issues with skinny callid
ASTERISK-21064: Crash when handling ACK on dialog that has no channel
ASTERISK-21065: Asterisk 11 IPv6 - FastAGI fail
ASTERISK-21066: Respect Callerid ID presentation
ASTERISK-21067: pointers of old channel_generator are not tidied up when a new channel_gernerator is activated
ASTERISK-21068: Asterisk is freezing (since 1.8.18.0 to 1.8.20.1) when doing 'core show channels' AND receiving 'SIP register'
ASTERISK-21069: xmpp distributed device states aggregation update fails
ASTERISK-21070: DBdeltree throws spurious error under almost all cases
ASTERISK-21071: [patch] Open channel for incoming call after RING; +CLIP responses from rfcomm; faster reporting of incoming calls
ASTERISK-21072: Implement directmedia in chan_gulp
ASTERISK-21074: Implement NAT settings in chan_gulp
ASTERISK-21076: Implement non-session based messaging support (RFC 3428)
ASTERISK-21077: Add support for video in chan_gulp
ASTERISK-21080: Redial button does not work properly
ASTERISK-21081: New SIP Channel Driver - Registrar - Part One
ASTERISK-21082: New SIP Channel Driver - Registrar - Part Two
ASTERISK-21083: New SIP Channel Driver - Registrar - Part Three
ASTERISK-21084: New SIP Channel Driver - Path Support
ASTERISK-21089: New SIP Channel Driver - Test Plan for Basic Calls
ASTERISK-21091: Add 0x144 skinny support
ASTERISK-21094: MixMonitorMute mutes through stream if already slinear (e.g. Originate)
ASTERISK-21095: More called details fixup
ASTERISK-21096: Complete channel snapshot work for Stasis Core
ASTERISK-21097: Stasis Core - Refactor MWI support
ASTERISK-21098: Asterisk 1.8.12.0 core dumps
ASTERISK-21099: Reload makes dahdi not work
ASTERISK-21101: Stasis Core - Refactor Device State support
ASTERISK-21102: Stasis Core - Refactor Presence State support
ASTERISK-21103: Stasis Core - Refactor the other event types onto the Stasis Core message bus
ASTERISK-21108: If chan_motif fails to load, Asterisk still thinks it's loaded
ASTERISK-21111: segfault Asterisk 1.6.0.9
ASTERISK-21113: app_dial.c does not honor 'c' flag when calling party hangs up
ASTERISK-21117: Bad interpretation of the file chan_dahdi.conf when using open r2 parameters
ASTERISK-21119: Asterisk system locks up with chan_unistim
ASTERISK-21120: Unable to properly hang up calls when second line rings
ASTERISK-21122: Documentation on the various methods of changing verbosity can be confusing
ASTERISK-21123: Compilation error: pjproject/build.mak: version.mak: No such file or directory
ASTERISK-21124: cdr_adaptative_odbc does not populate cdr information
ASTERISK-21125: Asterisk 11 needs libuuid in configure script due to pjproject
ASTERISK-21127: Empty custom CDR value
ASTERISK-21128: Locking inversion when attempting to set caller ID while holding iaxsl lock causes deadlock
ASTERISK-21129: Lock on do_monitor
ASTERISK-21130: sip_pvt.dsp incorrect manipulation related to inband dtmfmode and faxdetect in SIPDtmfMode() app. and enable_dsp_detect()
ASTERISK-21131: [patch] - Asterisk creates SDP with (peer) unsupported audio codec
ASTERISK-21133: SIP/TDM interworking, and RTP on CALL PROCEEDING
ASTERISK-21134: On Asterisk 11.2.1 can't load chan_dahdi.so
ASTERISK-21135: Asterisk 1.8 no longer sends unsolicited message-summary (NOTIFY) after realtime SIP peer registers
ASTERISK-21139: Asterisk 11 Seg Faults on READ after ConfBridge KICK
ASTERISK-21141: RPID not parsed correctly if display-name is *(token LWS)
ASTERISK-21142: Page app can't find ConfBridge
ASTERISK-21144: One way audio after channels are AMI Bridged out of a ConfBridge that has jitterbuffer=yes
ASTERISK-21145: asterisk doesn't continue to the next priority after a soft hangup
ASTERISK-21146: Semi attended (blonde) transfer causes queue member not to respect wrapuptime
ASTERISK-21148: [patch] - Asterisk use '(null)' in 'via' header and 'call-id' header when relaying SIP MESSAGE
ASTERISK-21149: detailed hangup cause for ${REASON} variable in call files
ASTERISK-21150: UDPTL Error Correction Scheme Negotiation Issue, Asterisk copies the Error Correction Scheme from T38 offer from remote peer even if UDPTL error correction scheme is set to NONE for that peer in sip.conf
ASTERISK-21151: 'Squelching' early media in DAHDI (sig_pri)
ASTERISK-21152: if pressed * the user menu is not working
ASTERISK-21153: Getting duplicate entry in CDR entry for same call
ASTERISK-21155: CallerID on Indian PSTN is not working.
ASTERISK-21156: Asterisk crashes with XMPP\Google Voice config where username is missing hostname portion
ASTERISK-21157: Asterisk 1.8.20.0 Crash when unloading chan_dahdi
ASTERISK-21158: Video enabled peers will send a video stream when calling a voice only peer.
ASTERISK-21160: In an XMPP distributed device state configuration, setting pubsub_node to a value without the pubsub prefix can cause a dialog loop leading to high CPU usage and crashiness
ASTERISK-21162: Deadlock in cdr.c: cdr_batch_lock vs cdr_pending_lock
ASTERISK-21163: pjproject raises an assert failure when creating TURN socket while adding ICE candidates to RTP session
ASTERISK-21164: Need clarification on distributed device state behavior and whether this behavior is a possible regression
ASTERISK-21168: asterisk logger stops logging VERBOSE and NOTICE messages after some time.
ASTERISK-21170: DTMF timestamp issue
ASTERISK-21172: One way audio when external Call forwarded to queue member
ASTERISK-21173: [patch] example sippeers sql hasn't been adapted for ipv6 and causes chan_sip to generate a warning message
ASTERISK-21174: Asterisk 11 auto-pause problem
ASTERISK-21176: Call files on OS X, using KQueue, do not get processed (load 100%)
ASTERISK-21177: [patch] Issues with skinny callinfo during fwd
ASTERISK-21178: Improve documentation for manager command Getvar, Setvar
ASTERISK-21180: Implement channel state events for Stasis HTTP
ASTERISK-21182: Create documentation/specification for expected Stasis HTTP events
ASTERISK-21184: chan_gulp: Add support for multiple media streams/types
ASTERISK-21186: chan_gulp - Implement media negotiation rules
ASTERISK-21190: chan_mgcp crash on chunked m= sdp line
ASTERISK-21191: [patch] - VoiceMailPlayMsg doesn't work with ODBC
ASTERISK-21193: IAX/2 fails to destroy channel on max retransmits exceeded
ASTERISK-21194: chan_sip can fail to find a peer during reload
ASTERISK-21195: Transcoding makes bad choice in high-rate translations
ASTERISK-21196: Refactor CDRs onto Stasis-Core to handle changes in bridging behavior
ASTERISK-21199: Implement outbound authentication handling support
ASTERISK-21201: [patch] In Manager Interface, SIP registry event does not show username on Status: Registered
ASTERISK-21202: Asterisk SIP message (SMS) stops working
ASTERISK-21203: res_xmpp socket error: takes upto 19 minutes to restore xmpp socket connection to google
ASTERISK-21204: Asterisk increments the session version in 2xx message even if a '183 Session in Progress' with SDP has already been sent in response to initial INVITE.
ASTERISK-21205: [patch] dundi_read_result crash due to negative number
ASTERISK-21206: Crashes contstantly in chan_motif
ASTERISK-21207: [patch] - Deadlock on fax extension calling ast_async_goto() with locked channel
ASTERISK-21208: 'sip set options {on|off}' command to explicitly enable 200 OK responses to OPTIONS
ASTERISK-21209: crash in res_clialiases on reload
ASTERISK-21210: BRI locks up
ASTERISK-21211: chan_iax2 - unprotected access of iaxs[peer->callno] potentially results in segfault
ASTERISK-21215: Unexpected Behavior in Adaptive CDR when U() used on Dial
ASTERISK-21216: Skinny voicemail indication issues
ASTERISK-21222: Behavior of 'logger set level' With Respect To Entries in logger.conf
ASTERISK-21223: Asterisk no longer responds to SIP REGISTER's that don't contain an Authorization
ASTERISK-21224: [patch] Skinny groupPickup issues
ASTERISK-21225: [patch] Setting nat=force_rport in [general] sip.conf will never work
ASTERISK-21226: one way audio after call was on hold
ASTERISK-21228: Deadlock in pbx_find_extension when attempting an autoservice stop due to holding the context lock
ASTERISK-21231: When outboundpoxy is set, asterisk should not attempt to resolve DNS
ASTERISK-21232: Asterisk sends AUDIO REINVITE when session timer expires in T38 call
ASTERISK-21233: [patch] Downgrade missing speeddials to a template debug and remove unsupported message for 7937
ASTERISK-21234: Deadlock when using two Local channels & fax gateway (local_queryoption)
ASTERISK-21236: 11.3 release
ASTERISK-21237: detect PRI_EVENT_NOTIFY(16) instead of PRI_EVENT_HANGUP_REQ(15)
ASTERISK-21241: When using voicemail as announce only (maxmsg=0), the star dtmf to enter the voicemail is not honored
ASTERISK-21242: Segfault when T.38 re-invite retransmission receives 200 OK
ASTERISK-21243: [patch] Backport Appropiate NAT Setting Cleanups To 1.8
ASTERISK-21244: Crash in libsrtp when attempting to unprotect RTCP packet
ASTERISK-21245: Application Dial - Option g ignored with option A if the called part hangs up during the announce.
ASTERISK-21246: [patch] use of rtpkeepalive uses CN packet with marker bit set, plus a ULAW payload instead of CN
ASTERISK-21248: CALLERID(dnid-num-plan) does not get any value set.
ASTERISK-21250: AddQueueMember does not write membername to the queue log
ASTERISK-21253: Create a test realtime backend suitable for driving Asterisk Test Suite tests
ASTERISK-21255: Create a sorcery wizard for the AstDB
ASTERISK-21257: Implement inbound/outbound Caller ID handling
ASTERISK-21258: Implement mid-call connected line support for chan_gulp
ASTERISK-21259: Build a pub/sub architecture for the new SIP channel driver
ASTERISK-21260: Add MWI support to the new SIP channel driver
ASTERISK-21261: Add DTMF Info support
ASTERISK-21267: Add stasis-http configuration
ASTERISK-21270: Bridge API Enhancements - add subclassing ability to ast_bridge
ASTERISK-21271: Bridge API Enhancements - subclass ConfBridge with its own Virtual Method table
ASTERISK-21272: Bridge API Enhancements - subclass Parking with its own Virtual Method table
ASTERISK-21277: stasis-http authentication
ASTERISK-21278: stasis-http Cross-Origin configuration
ASTERISK-21279: Allow WebSocket connections on URL's other than /ws
ASTERISK-21280: Basic configuration for stasis-core
ASTERISK-21281: stasis-http: Create Confluence swagger-codegen templates
ASTERISK-21282: Add DTMF events to the stasis-http WebSocket
ASTERISK-21283: Implement stasis-http POST /api/channels/{channelId}/play
ASTERISK-21292: Add callfwd_noanswer to skinny
ASTERISK-21293: Script to calculate number of each character sequence occurrence specified as an argument
ASTERISK-21294: Calling StopMixMonitor on a channel w/o MixMonitor running returns -1
ASTERISK-21295: Sip registration fails, wrong parsing when secret has parentheses symbol
ASTERISK-21296: SIP module not responding any more
ASTERISK-21297: Segmentation fault on hangup in in ast_bridged_channel
ASTERISK-21298: Confbridge recording fails - deadlock
ASTERISK-21299: Asterisk send wrong codec order in the leg B of the call
ASTERISK-21300: Asterisk is sending wrong codec order in the leg B of the call
ASTERISK-21301: ERROR and failure to resolve socket address due to whitespace after port number in SIP Via header
ASTERISK-21302: [patch] app_voicemail crashes on config error and there are some potential memory leaks
ASTERISK-21303: qualifygap SIP general setting appears broken
ASTERISK-21304: [patch] AGI AsyncAGI event returns AGI command arguments
ASTERISK-21305: Segfault when hanging up channels active in MeetMe with recording
ASTERISK-21306: set FEATURE(parkingtime) is not inherited by child channels
ASTERISK-21310: __sip_xmit fails with interrupted system call
ASTERISK-21311: CLI command 'module load' attempts to free unallocated memory on tab completion
ASTERISK-21314: Sip channel is deadlocked
ASTERISK-21315: Asterisk Realtime using res_config_mysql not recognizing port field for SIP peers
ASTERISK-21316: Segfault on ast_channel_tech(chan)->send_digit_begin
ASTERISK-21318: AMI events for ConfBridge Mute/Unmute and Record start/stop
ASTERISK-21320: fails to parse irregular version strings (such as ~dfsg)
ASTERISK-21321: Skinny softkey endcall when transferring should not blind xfer
ASTERISK-21322: fails to copy relative symlinks from the tree
ASTERISK-21323: Asterisk 11 svn branch and srtp - white noise only
ASTERISK-21324: [patch] Per-user option 'allowmultiplelogin' in manager
ASTERISK-21325: astdb2sqlite is very slow when running the 32-bit version on 64 system
ASTERISK-21327: Add transfer softkey when transferor chan ringing
ASTERISK-21328: ISDN PRI Release Delay
ASTERISK-21329: chan_alsa: patch for crash when audio device in unexpected state
ASTERISK-21330: XML documentation generation fails on fresh checkouts
ASTERISK-21331: Bridge API Enhancements - add bridging unique identifier, bridge container, and basic CLI commands
ASTERISK-21332: Bridge API Enhancements - create the Basic Bridge subclass
ASTERISK-21333: Bridge API Enhancements - refactor all uses of a jitter buffer to use func_jitterbuffer
ASTERISK-21334: Bridge API Enhancements - hide masquerades
ASTERISK-21335: Bridge API Enhancements - add externally initiated blind transfers
ASTERISK-21336: Bridge API Enhancements - add externally initiated attended transfers
ASTERISK-21337: Bridge API Enhancements - add stasis core messages for blind/attended transfers
ASTERISK-21338: Bridge API Enhancements - Refactor the Dial API as a bridge mixing technology
ASTERISK-21339: Bridge API Enhancements - add CCSS, Connected Line, Pre-Dial to the Dial API
ASTERISK-21352: Bridge API Enhancements - refactor ParkAndAnnounce application to use the new parking bridge
ASTERISK-21353: Bridge API Enhancements - add features to parking
ASTERISK-21354: Bridge API Enhancements - perform basic parking pickup
ASTERISK-21356: Segfault during bridge channel proxy inspection in a masquerade caused by an AMI Redirect of two channels
ASTERISK-21359: Refactor AMI DTMF events onto Stasis-Core
ASTERISK-21366: Transfer settings to compile asterisk menuselect.makeopts copying from old installation
ASTERISK-21367: Executing StopMixMonitor on channel is not MixMonitored hangs up this channel
ASTERISK-21368: Add Manager Events for SIP Registry status changing
ASTERISK-21369: Need to INVITE to peer with other domain without peer domain addition
ASTERISK-21370: Call gets dropped transferring to external destination
ASTERISK-21373: In proxy NAT traversal situation the far end requires Asterisk to send RTP first
ASTERISK-21374: [patch] One-way Audio With auto_* NAT Settings When SIP Calls Initiated By PBX
ASTERISK-21377: After sip reload 80% peers is UNREACHEBLE
ASTERISK-21378: chan_sip completely blocks on DNS lookups
ASTERISK-21382: make (at least some of) the payload numbers used by dynamic payload types configurable
ASTERISK-21383: STUN Binding Requests Not Being Sent Back from Asterisk to Chrome
ASTERISK-21384: Unique ID Call Count Increasing By 2
ASTERISK-21385: SIP Channel Lock
ASTERISK-21386: SIP Channel Locks
ASTERISK-21387: Asterisk unresponsive while waiting for lock MUTEX 6239 __ast_pbx_run c
ASTERISK-21389: res_timing_pthread fails to return from write, causing timer dependent operations to block indefinitely
ASTERISK-21390: New applications for app_stack.c: GosubEntry and StackPopGoto
ASTERISK-21391: Asterisk fails to load chan_sip.so after compiling with "#define REF_DEBUG 1" in chan_sip.c
ASTERISK-21394: [patch] - Fundamental changes to CDR within single asterisk family (1.8) during externally initiated blind transfers with an h extension present
ASTERISK-21395: High CPU use due to invalid time limit passed to kevent() in pbx_spool.c:scanthread
ASTERISK-21397: [patch] manager crash on unloading app_queue
ASTERISK-21398: [patch] chan_iax2.c:7998 authenticate_verify: requested inkey 'my_oth' for RSA authentication does not exist
ASTERISK-21399: RTP Multicast of L16 (type 10): Asterisk and wireshark disagree
ASTERISK-21400: Websocket Call Ends in Match Not Found
ASTERISK-21401: [patch] codec_resample cannot be unloaded
ASTERISK-21402: [patch] Unloading app_queue can cause AMI to segfault
ASTERISK-21406: [patch] chan_sip deadlock on monlock between unload_module and do_monitor
ASTERISK-21407: [patch] features_shutdown doesn't finish cleanup
ASTERISK-21409: [patch] - Race condition with IAX2 transfer, 2 releases happen on same call legs. locks up with many threads blocked by iax2_destroy_helper
ASTERISK-21410: Park() application never returns in some cases
ASTERISK-21411: Far end Re-invites Asterisk without SDP (ACKs with SDP) - Asterisk does not modify port of RTP stream - Ignores ACK due to no SDP version change
ASTERISK-21412: [patch] config.c/config_text_file_load() leaks globbuf
ASTERISK-21413: app_voicemail sound file for forwarding messages can be misleading
ASTERISK-21414: Recording with MixMonitor consumes more G.729 licenses than expected on a single call
ASTERISK-21415: Asterisk Segmentation Fault when reloading module a few times in a short delay
ASTERISK-21416: Implement SDES-SRTP support in chan_gulp
ASTERISK-21419: Implement DTLS-SRTP support in chan_gulp
ASTERISK-21421: API Improvements: build out the concept of an endpoint in Stasis-Core
ASTERISK-21422: Asterisk Test Suite - rework our CEL testing module to be reliable
ASTERISK-21424: Implement chan_gulp tests - off nominal incoming call paths
ASTERISK-21426: New SIP Channel Driver - Call Forwarding
ASTERISK-21429: Distributed Device State using JABBER/XMPP not working since Secuity Advisory AST-2012-015
ASTERISK-21430: [patch] Call ID missing when logging through syslog
ASTERISK-21432: Video isn't negotiated when endpoint switches to video on an established SIP to SIP call
ASTERISK-21433: Add analogous support for 'alwaysauthreject' to chan_gulp and top level security settings
ASTERISK-21434: Add anonymous access support to chan_gulp
ASTERISK-21435: Add redirecting information support to chan_gulp
ASTERISK-21436: Add CLI/AMI initiated NOTIFY requests (sip_notify support)
ASTERISK-21441: New SIP Channel Driver: Create an API on top of the pub/sub framework for extension state notifications
ASTERISK-21442: New SIP Channel Driver - Create an extension state provider for RFC 3863
ASTERISK-21443: New SIP Channel Driver - Create a state provider for dialog-info+xml
ASTERISK-21447: Asterisk crashes while connecting to TCP peers
ASTERISK-21448: New SIP Channel Driver - basic fax support
ASTERISK-21450: Allow pluggable modules to be executed against particular Asterisk Versions
ASTERISK-21452: New SIP Channel Driver - Create Event State Compistor resource module and implement Publish API
ASTERISK-21453: New SIP Channel Driver - Implement CCSS
ASTERISK-21456: New SIP Channel Driver - add basic REFER support and SIP blind transfers
ASTERISK-21457: New SIP Channel Driver - enhance basic REFER support to handle SIP attended transfers
ASTERISK-21460: New SIP Channel Driver - create a SIP Security Event module suitable for consumption in the new SIP stack
ASTERISK-21462: Stasis Core - Refactor random AMI events
ASTERISK-21464: with directrtpsetup some payload type identifiers from A party's INVITE are not copied to the INVITE for B party
ASTERISK-21465: wrong routing of ACK request following a 200OK (Record-Route header not taken into account)
ASTERISK-21466: [patch] [crash] command (sip show peers) crashes Asterisk with ~3500 registered peers
ASTERISK-21467: Stasis Core - Refactor MeetMe Events
ASTERISK-21468: Stasis Core - Refactor ConfBridge Events
ASTERISK-21469: Stasis Core - Refactor Queue Events
ASTERISK-21470: Stasis Core - Refactor AGI Events
ASTERISK-21471: Stasis Core - Refactor RTP/RTCP Events
ASTERISK-21472: Stasis Core - Refactor AOC Events
ASTERISK-21473: Stasis Core - Refactor CCSS events to Stasis-Core
ASTERISK-21474: Stasis Core - Refactor AddOn Channels
ASTERISK-21475: CallerID information doesn't persist after a Channel Redirect on a H323 leg (works with other channel technologies)
ASTERISK-21476: Stasis Core - Refactor extraneous channel events
ASTERISK-21486: Call answered by a dynamic agent and then SIP transferred to an external number is not written to CDR
ASTERISK-21487: Stasis Core - Refactor Hold event from chan_sip/chan_iax2/sig_pri to channel core
ASTERISK-21488: Stasis Core - Refactor Registry events from chan_iax2/chan_sip
ASTERISK-21489: Stasis Core - Refactor PeerStatus events
ASTERISK-21492: RTP packetization negotiated at ptime=30 for first leg; ptime=20 for second leg results in deltas of 20/40ms and 30/0ms
ASTERISK-21494: AMI 1.4 Improvements - Add a field to all AMI events that conveys the system name
ASTERISK-21495: 302 Moved Temporarily CDR Incorrect
ASTERISK-21496: Stasis Core - Add the Transfer bridging message and corresponding AMI event
ASTERISK-21499: New SIP Channel Driver - add/finish up SIP Qualify Support
ASTERISK-21500: New SIP Channel Driver - Add provisional keep alives
ASTERISK-21501: New SIP Channel Driver - add custom INFO support for one touch recording
ASTERISK-21502: New SIP Channel Driver - add Advice of Charge support
ASTERISK-21503: New SIP Channel Driver - integrate stasis endpoints
ASTERISK-21504: New SIP Channel Driver - add SIP History tracking
ASTERISK-21505: New SIP Channel Driver - add call pickup group configuration options
ASTERISK-21506: New SIP Channel Driver - add a variety of customization configuration parameters
ASTERISK-21507: New SIP Channel Driver - add progressinband
ASTERISK-21517: API Improvements: refactor app_queue to listen for a Transfer stasis message and update the Queue Log appropriately
ASTERISK-21518: Bridge API Enhancements - refactor chan_iax2 to perform blind transfers using the new bridging framework
ASTERISK-21519: Bridge API Enhancements - implement blind transfers in chan_sip
ASTERISK-21520: Bridge API Enhancements - implement attended transfers in chan_sip
ASTERISK-21522: [patch] DTMF end is not always processed, causes one-way audio
ASTERISK-21523: Bridge API Enhancements - refactor sig_pri_attempt_transfer to use Bridging Framework
ASTERISK-21524: Bridge API Enhancements - refactor chan_misdn's misdn_attempt_transfer
ASTERISK-21525: Bridge API Enhancements - refactor chan_mgcp attempt_transfer
ASTERISK-21526: Bridge API Enhancements - refactor chan_skinny skinny_transfer
ASTERISK-21527: Bridge API Enhancements - refactor chan_unistim attempt_transfer
ASTERISK-21542: Bridge API Enhancements - get DTMF attended transfers feature complete - configuration support
ASTERISK-21543: Bridge API Enhancements - get DTMF attended transfers feature complete - add attended transfer monitoring
ASTERISK-21544: Bridge API Enhancements - get call pickup working
ASTERISK-21549: AMI 1.4 Improvements - refactor ast_pbx_outgoing_* to use the dial API; add Originate AMI Events
ASTERISK-21550: AMI 1.4 Improvements - Add Dial Begin/End messages to FollowMe
ASTERISK-21551: AMI 1.4 Improvements - Add Dial Begin/End messages to Queue
ASTERISK-21552: AMI 1.4 Improvements - Refactor AMI/CLI channel inspection actions to use the stasis channel cache
ASTERISK-21553: Bridge API Enhancements - add one touch recording
ASTERISK-21554: Bridge API Enhancement - do something about chan_agent
ASTERISK-21555: Bridge API Enhancements - implement channel variables in the bridging core
ASTERISK-21563: API Enhancements - CEL refactoring - channel state
ASTERISK-21564: API Enhancements - CEL refactoring - bridge state
ASTERISK-21565: API Enhancements - CEL refactoring - transfers
ASTERISK-21566: API Enhancements - CEL refactoring - cleanup
ASTERISK-21567: API Enhancements - CEL refactoring - Documentation
ASTERISK-21573: CallCompletionRequest() not available with call forwarding disabled on busy extension
ASTERISK-21574: Queue is sending multiple calls to the available agents at once when autofill is enabled
ASTERISK-21575: Asterisk REST API - Implement GET /asterisk/info call
ASTERISK-21576: Asterisk REST API - Implement /recordings
ASTERISK-21577: Asterisk REST API - Update the /recording template
ASTERISK-21578: Asterisk REST API - Create the sounds resource template
ASTERISK-21579: Asterisk REST API - Create the playback template
ASTERISK-21580: Asterisk REST API - Update bridge/channel templates for playback and record operations
ASTERISK-21581: Asterisk REST API - Implement GET /recording/{id}
ASTERISK-21582: Asterisk REST API - Implement DELETE /recording/{id}
ASTERISK-21583: Asterisk REST API - Implement POST /recording/{id}/rename
ASTERISK-21584: Asterisk REST API - Implement GET /sounds
ASTERISK-21585: Asterisk REST API - Implement GET /sound/{id}
ASTERISK-21586: Asterisk REST API - implement GET /playback/{id}
ASTERISK-21587: Asterisk REST API - Implement POST /playback/{id}/control
ASTERISK-21588: Asterisk REST API - Implement POST /playback/{id}/restart
ASTERISK-21589: Asterisk REST API - Implement POST /playback/{id}/pause
ASTERISK-21590: Asterisk REST API - Implement POST /playback/{id}/rewind
ASTERISK-21591: Asterisk REST API - Implement POST /playback/{id}/fastforward
ASTERISK-21592: Asterisk REST API - Implement POST /bridge/{id}/play
ASTERISK-21593: Asterisk REST API - Implement POST /bridge/{id}/record
ASTERISK-21594: Asterisk REST API - Implement POST /channel/{id}/record
ASTERISK-21615: Asterisk REST API - Implement GET /endpoints
ASTERISK-21616: Asterisk REST API - Implement GET /endpoint/{id}
ASTERISK-21617: Asterisk REST API - Implement POST /channels to an endpoint
ASTERISK-21618: Asterisk REST API - Implement POST /channels/{id}/mute and /channels/{id}/unmute
ASTERISK-21619: Asterisk REST API - Implement POST /channel/{id}/hold and /channel/{id}/unhold
ASTERISK-21620: Asterisk REST API - Implement POST /channel/{id}/dial
ASTERISK-21621: Asterisk REST API - Implement GET /bridges
ASTERISK-21622: Asterisk REST API - Implement GET /bridge/{id}
ASTERISK-21623: Asterisk REST API - Implement DELETE /bridge/{id}
ASTERISK-21624: Asterisk REST API - Implement POST /bridges
ASTERISK-21625: Asterisk REST API - Implement POST /bridge/{id}/addChannel
ASTERISK-21626: Asterisk REST API - Implement POST /bridge/{id}/removeChannel
ASTERISK-21639: Segfault in app_confbridge while stress testing
ASTERISK-21640: Bridge API Enhancements - work through a channel being removed from a bridge by an external party
ASTERISK-21641: Bridge API Enhancements - get Park AMI action working again
ASTERISK-21642: Bridge API Enhancements - add hints back to the parking slots
ASTERISK-21643: Bridge API Enhancements - add the default parking lot
ASTERISK-21644: Bridge API Enhancements - add dynamic parking lots
ASTERISK-21645: Bridge API Enhancements - add parking dialplan generation
ASTERISK-21654: DNS SRV lookup doesn't bother with family (ipv4, ipv6)
ASTERISK-21657: asterisk locks up after running traffic
ASTERISK-21658: Asterisk REST API - Implement POST /channels to a dialplan context/extension/priority
ASTERISK-21660: Queue does not respect agent status - deliver a call even agent is busy or unavailable
ASTERISK-21661: SMS delvery to VoIP mobile
ASTERISK-21662: Res_odbc keeps losing connection to MySQL
ASTERISK-21663: [patch] Realtime TCP endpoints lose registration after "sip reload" & "core reload"
ASTERISK-21664: Asterisk terminates calls if Session-Expires isn't present on INVITE
ASTERISK-21665: 11.X Crash on debian/sparc with SIGBUS, Bus Error
ASTERISK-21666: patch to implement match_auth_username option(sip.conf) for SIP REGISTER
ASTERISK-21667: No AMI events output until Asterisk receives an AMI command
ASTERISK-21668: Basic res_sip XML documentation
ASTERISK-21669: Fix dependencies on res_sip files
ASTERISK-21670: Coding style within chan_gulp
ASTERISK-21671: Asterisk realtime/ SIP status
ASTERISK-21672: Early media not properly handled on outbound TCP trunk
ASTERISK-21673: Asterisk conversion of 183 without SDP to 180 breaks interoperability with Microsoft Lync
ASTERISK-21675: Asterisk forgot G729 license
ASTERISK-21676: (CALLERPRES()=prohib) not honoured over ISDN
ASTERISK-21677: NOTIFYs for BLF start queuing up and fail to be sent out
ASTERISK-21678: IPv6-configured sip channel transmits to (null) for IPv4 register= hosts
ASTERISK-21683: Asterisk 1.8.21.0 Blind Transfer To Parking For An Inbound Call Fails And Leaves Call In Limbo State
ASTERISK-21688: CDR record cannot be modified in 'h' extension when 'g' option is used in Dial application
ASTERISK-21689: AMI bridge continues to try and bridge even after reporting "Channel2" does not exist; and fails resulting in hangup of Channel1
ASTERISK-21690: Asterisk sends SIP 481 after REFER
ASTERISK-21691: bridge cmd does not hangup if the bridged leg hangedup
ASTERISK-21693: Use of possibly uninitialized value in ast_channel_hangupcause_hash_set
ASTERISK-21694: Peer with outbound proxy is resolved in DNS
ASTERISK-21695: Crash Asterisk
ASTERISK-21696: Assertion error results in crash in pjproject's ICE worker thread
ASTERISK-21697: Bridge API Enhancements - handle Local Channel Optimization in CDRs
ASTERISK-21698: Bridge API Enhancements - handle Attended Transfers in CDRs
ASTERISK-21699: Bridge API Enhancements - handle Call Pickup in CDRs
ASTERISK-21703: New SIP Channel Driver - write a registration test plan on the wiki
ASTERISK-21708: Bridge API Enhancements - write a test plan for Queues
ASTERISK-21710: New SIP Channel Driver - implement the promiscredir option in chan_gulp
ASTERISK-21711: Stasis API - Incorporate the bridging framework into res_stasis app
ASTERISK-21713: Bridge API Enhancements - Create a media channel for the bridging API
ASTERISK-21716: [patch] logger thread sometimes exits with messages still queued
ASTERISK-21717: [patch] - Documentation for PASSTHRU function is unclear
ASTERISK-21718: [patch] pbx_dundi leaks ast_io_add
ASTERISK-21719: [patch] res_srtp doesn't cleanup srtp library
ASTERISK-21720: Asterisk 11 cannot compile with multiple definitions. Possible libasteriskssl + openssl issue.
ASTERISK-21721: SIP Failed to parse multiple Supported: headers
ASTERISK-21722: chan motif behaves wrong
ASTERISK-21723: [patch] pbx cleanup is incomplete
ASTERISK-21724: [patch] __ast_rwlock_destroy can segfault with DEBUG_THREADS
ASTERISK-21725: Asterisk 11 attempts IPv6 (with an insane address) when talking to an IPv4-only endpoint
ASTERISK-21726: Asterisk does not properly parse multiple allow: headers
ASTERISK-21737: [patch] - Crash during transfer from DAHDI/SIP to SIP/SIP in ast_format_cap_append called from remote bridge loop
ASTERISK-21738: [patch] Segfault On Realtime Queue Members Processing
ASTERISK-21741: [patch] - Improved Caller ID Diagnostics and Processing for FXO Channels
ASTERISK-21742: SIP Session-Expires: Set timer to correctly expire at (~2/3) of the expiry interval when not the refresher.
ASTERISK-21743: [patch] - Core show Locks, Include Asterisk version.
ASTERISK-21744: [patch] - fix lower bound check with -ve integer conversion from a float
ASTERISK-21751: Asterisk crashes with segmentation fault when trying to do a pickup with INVITE with Replaces
ASTERISK-21752: Asterisk peers with host=<dnsname> do not accept calls from all hosts in dnsname's multiple host SRV record set
ASTERISK-21753: Seg Fault while attempting to queue AST_CONTROL_SRCCHANGE on a NULl channel when handling an incoming SIP ACK over TCP
ASTERISK-21754: Bridge API Enhancements - write tests for the new Bridge AMI actions/events
ASTERISK-21756: assert() when using dtmfmode=none
ASTERISK-21757: segfault on asterisk startup: motif iksemel
ASTERISK-21758: chan_gtalk and res_xmpp not compiling with openssl-devel installed if iksemel-devel is not
ASTERISK-21760: Asterisk autoconf script does not check for pkg-config as a dependency
ASTERISK-21761: sip call stuck, crash
ASTERISK-21762: IAX2 call problem
ASTERISK-21763: asterisk -r Bus Error on Debian/sparc
ASTERISK-21765: [patch] - FILE function's length argument counts from beginning of file rather than the offset
ASTERISK-21768: LICENSE file missing from Asterisk Extra sounds (French)
ASTERISK-21772: Redundant if statement in dns.c
ASTERISK-21773: Asterisk 1.8.22.0 Open Blockers
ASTERISK-21774: Asterisk 11.4.0 Open Blockers
ASTERISK-21775: FPE during MOH playback
ASTERISK-21777: Asterisk tries to transcode video instead of audio
ASTERISK-21778: astobj2.c:115 INTERNAL_OBJ: user_data is NULL followed by Segmentation fault on cancelled divert
ASTERISK-21779: Manager closes connection when a SendText action is requested during hangup
ASTERISK-21780: Add missing documentation for new config option
ASTERISK-21781: on reload app_queue should check/prune queue members against associated devices in configuration
ASTERISK-21782: Delayed audio to agent when answering a queue call
ASTERISK-21785: __ao2_ref_debug() logs to /tmp/refs when REF_DEBUG is not defined
ASTERISK-21786: Segfault in MyODBC MySQL connector during reconnect attempt when connection is lost
ASTERISK-21787: No IAX2 communication either user/peer or friend accounts
ASTERISK-21788: What variable stores caller queue position
ASTERISK-21789: ast_http_get_cookies() fails in the presence of RFC2965 Cookie2 header
ASTERISK-21792: chan_sip.c: Autodestruct on dialog X with owner X in place (Method: BYE). Rescheduling destruction for X
ASTERISK-21793: Segmentation fault when dealing with Agent channels
ASTERISK-21794: CLI command 'realtime update2' syntax failure when using according to usage help
ASTERISK-21795: failed compilation - dns.c references res_nsearch which is not available on uclibc
ASTERISK-21797: FILE function reads inconsistently
ASTERISK-21799: [patch] Dropouts/distortion in MixMonitor recording when recording RTP with ptime of 60ms
ASTERISK-21800: ooh323 channels stuck if no gatekeer or ooh323 reload
ASTERISK-21802: (un)muting a ConfBridge user via *CLI doesn't generate AMI events
ASTERISK-21803: transcoding from silk to g711 constantly print the message "lintosilk_frameout: encoding XXX samples"
ASTERISK-21804: CLI command 'reload' has inconsistent output written to console when operating with a DEBUG level greater than 0
ASTERISK-21807: Wrong reference and missing values for "Hangup Hause Code Mappings"
ASTERISK-21809: [patch] sip_pvt members novideo and notext are being reset to TRUE every time SDP is processed
ASTERISK-21811: cdr_odbc "CDR direct execute failed" Not working with MySQL Master Server
ASTERISK-21812: Whitespace not escaped correctly on AGI()
ASTERISK-21814: "/" in TOUCH_MIXMONITOR is replaced by "-"
ASTERISK-21815: SipRemoveHeader does not remove previously added Alert-Info Header
ASTERISK-21816: [patch] OpenBSD fix for UUID
ASTERISK-21817: Stasis-HTTP: Implement Stasis message_type formatting functions
ASTERISK-21818: Stasis-HTTP: Write a python module that provides testing functionality around the REST API in the Test Suite
ASTERISK-21822: Adaptive CDR Not Written When Call Ends In Blind Transfer
ASTERISK-21823: Segfault when autoload=yes in modules.conf
ASTERISK-21824: SIP INVITES from Telphin broke in 1.8.21
ASTERISK-21825: [patch] websocket segmentation fault on certain invalid input
ASTERISK-21826: [patch] Bad queue_log entry when removed member from queue via CLI
ASTERISK-21827: [patch] Add kick all capability to app_confbridge's CLI command 'kick'
ASTERISK-21828: [patch] app_meetme.so hints load as Unavailable instead of Idle on start up
ASTERISK-21829: Bridge API Enhancements - finish connected line/redirecting handling in the bridging core
ASTERISK-21831: Fix skipped cdr/blind-transfer-accountcode test for 12
ASTERISK-21833: New SIP Channel Driver - implement nominal registration tests - test 1
ASTERISK-21834: New SIP Channel Driver - implement nominal registration tests - tests 2 and 3
ASTERISK-21835: New SIP Channel Driver - implement nominal registration tests - test 4, 5, and 6
ASTERISK-21836: New SIP Channel Driver - implement off-nominal registration tests
ASTERISK-21837: New SIP Channel Driver - implement nominal and off-nominal unregister tests
ASTERISK-21843: Failed Dial() in a call file does not post a CDR record
ASTERISK-21845: maxcalls exceeded, Asterisk sends out 480 and also BYE
ASTERISK-21846: RINGNOANSWER event for an agent in queue, but data1 field is null
ASTERISK-21847: Segfault due to dahdi_restart and round robin
ASTERISK-21849: Transfer Asterisk queues are not seen in cdr reports
ASTERISK-21854: Long Asterisk-version strings display improperly in the 'Connected to ...' line upon remote console connection
ASTERISK-21855: Asterisk crashes when XMPP message is sent (JabberSend) and no internet connection is available
ASTERISK-21856: STUN never works when asterisk started without internet access
ASTERISK-21857: Move Stasis-HTTP websocket URL from /ws to /stasis/events
ASTERISK-21859: Confbridge doesn't tear down an empty conference bridge when all users were kicked via end_marked=yes. Also, side effect crashes.
ASTERISK-21862: Add manager commands
ASTERISK-21863: [patch] RTP Native Bridge Codec Change Handling - Appears to compare immediately after setting equal
ASTERISK-21865: STRFTIME sometimes returns '0000-00-00 00:00:00'
ASTERISK-21867: IAX Trunk Realtime - ipaddr gets set to (null) in iax_trunk table
ASTERISK-21868: Asterisk REST API - Implement channel variables/global variables
ASTERISK-21870: Asterisk REST API - Add dialplan location to the 'release back to dialplan command'
ASTERISK-21872: high CPU usage ~15 seconds into call if rtpkeepalive set on channels when Asterisk is in a generic bridge and passing RFC2833 DTMF
ASTERISK-21873: Asterisk API Improvements - filter channels that should never be shown
ASTERISK-21875: Bridge API Enhancements - add CHANNEL(after-bridge-goto) feature
ASTERISK-21876: Bridge API Enhancements - add CHANNEL(dtmf-features)=[tkhwx] feature
ASTERISK-21877: Bridge API Enhancements - fix the Parking BUGBUG comments in trunk
ASTERISK-21879: app_queue's autofill=yes effectively fails to deliver all calls when those calls are preceded by a call with a min/max penalty that can't be delivered
ASTERISK-21882: Bridge API Enhancements - ensure that n-1 channels leaving a multi-party bridge ejects the last channel
ASTERISK-21883: Asterisk API Improvements - refactor channel/bridge inspection commands to query the Stasis Cache
ASTERISK-21884: Asterisk API Improvements - add AMI/CLI command for querying Stasis endpoints
ASTERISK-21885: Asterisk REST API - modify JSON events to include an event type field; update swagger and code generation to use a discriminated union
ASTERISK-21886: Bridge API Enhancements - add native bridging capabilities back to chan_dahdi
ASTERISK-21892: Segfault after fax
ASTERISK-21893: Segfault after call hangup, in ast_channel_hangupcause_set, at channel_internal_api.c
ASTERISK-21894: [patch] Initial support for SIP/TLS tlsverifyclient
ASTERISK-21895: Aborted in ast_hangup -> __ast_manager_event_multichan -> append_event
ASTERISK-21896: Aborted in ast_bridge_call -> ast_cdr_dup_unique_swap -> ast_cdr_dup
ASTERISK-21897: D option in Dial doesn't recognize "w" as a pause
ASTERISK-21898: Read application does not set the variable
ASTERISK-21899: Q850 Reason not forwarded between call legs
ASTERISK-21900: help text syntax example for Read application delivered onto wiki incorrectly
ASTERISK-21901: speex16 call to app_record with wav format results in a playable, but horrible sounding audio file
ASTERISK-21902: Configuring asterisk to use a certificate generated by a peer
ASTERISK-21903: [patch] Return proper result upon error when running some AGI commands
ASTERISK-21904: Whisper problem when the channel between Agent and client is in hold
ASTERISK-21905: Slow CODEC Translation compaired to the previous installation
ASTERISK-21906: [patch] Fix memory leaks, invalid reads and more reported by valgrind
ASTERISK-21907: Crash - segfault - When executing a MeetMeAdmin command that requires a member, without specifying a member
ASTERISK-21908: Asterisk do not log source IP for Fake auth rejection
ASTERISK-21911: Tearing down a registration throws a 403 back at the endpoint
ASTERISK-21912: Call hang-up when issuing mixmonitor start
ASTERISK-21913: Successive NOTIFY for MWI subscriptions isn't sent
ASTERISK-21916: Call hangs when FILTER function is used in dial plan
ASTERISK-21917: [patch] STUN crashes when SIP is bound to ipv4 and ipv6
ASTERISK-21918: WaitExten(5,m(default)) broken IAX2 channel
ASTERISK-21919: Originate request cause an INVITE from "anonymous.invalid" domain
ASTERISK-21920: IAX trunk timestamps set to zero when it's bridged SIP channel wraps RTP timestamps
ASTERISK-21921: Verbose() ignores remote console VERBOSE message verbosity
ASTERISK-21922: Add the ability to app_bridgwait to specify a particular bridge to place channels into
ASTERISK-21923: Add the ability to app_bridgewait to specify various music and sound options
ASTERISK-21924: Have the core bridging layer set the channel hang up cause on the channel/peer when the peer/channel breaks the bridge
ASTERISK-21925: Clean up the parking API in res_parking
ASTERISK-21930: [patch]WebRTC over WSS is not working.
ASTERISK-21931: Make menuselect displays warning that Makefile has a modification time in the future
ASTERISK-21932: [patch] ast_tls_cert: don't re-create generated files
ASTERISK-21933: DTMF feature hook triggered for both caller and peer when peer initiates blind transfer
ASTERISK-21938: PickupChan picks up answered call
ASTERISK-21939: New SIP Channel Driver - add CLI/AMI commands that force actions
ASTERISK-21941: menuselect possibly lies about speex dependencies
ASTERISK-21943: Bridge API Enhancements - handle AgentLogin/AgentLogout in the Queue Log using Stasis
ASTERISK-21944: Bridge API Enhancements - implement IAX2 native bridging (again)
ASTERISK-21947: New SIP Channel Driver - use the proper bridging API function to get the bridged channel during direct media tests
ASTERISK-21951: res_fax unsafely unlocks channel to perform an asynchronous goto to the 'fax' extension
ASTERISK-21953: connectedline parameter not documented
ASTERISK-21954: Local channel optimization needs to take into account frame hooks on the local channels.
ASTERISK-21955: [patch] Asterisk responds 488 to session timers re-Invite in an active T.38 dialog after exactly 5 seconds
ASTERISK-21956: MusicOnHold RealTime does not acknowledge announcement field
ASTERISK-21959: SPAMMY = NOTICE[24695]: chan_sip.c:27899 handle_request_subscribe: Received SIP subscribe for peer without mailbox:
ASTERISK-21960: ooh323 channels stuck
ASTERISK-21963: CLONE - Callers on Queue are not being delivered to free agents
ASTERISK-21964: SIP TLS Register statement fails if sip.conf register directive uses peer name.
ASTERISK-21965: [patch] Bug-fixed version of safe_asterisk not installed over old version
ASTERISK-21966: When FXS detects a fax, we should also disable callwaiting
ASTERISK-21967: CFLAG Improvement to prevent compiler error in Virtual Machine environments
ASTERISK-21968: Remove parkinglot from channel snapshots
ASTERISK-21969: Odd events during Stasis origination
ASTERISK-21970: Reconnects to an ARI websocket do not convey events for channels already in the application
ASTERISK-21971: POST to /stasis/bridges/?type=[bridgetype] fails silently
ASTERISK-21972: Bridge creation should allow for something more than mixing type during creation
ASTERISK-21973: ARI /bridges/{}/addChannel should allow an optional parameter specifying a role
ASTERISK-21974: ARI: Channels/bridges need MoH
ASTERISK-21976: Set more than one codec in dialplan execution using SIP_CODEC (adapted chan_sip:try_suggested_codec)
ASTERISK-21977: Stop potential message ordering issues between bridge and channel manager events
ASTERISK-21978: Crash caused by RAII_VAR in test_json when loading module
ASTERISK-21980: Error message for QUEUE_MEMBER when member is not in queue is unclear
ASTERISK-21981: Pass-through support for Opus and VP8 formats
ASTERISK-21991: [patch] - install a systemd service unit
ASTERISK-21996: chan_iax2 fails to process network packets after a while
ASTERISK-21997: [patch] - Incorrect Ring tone for Malaysia
ASTERISK-21998: ChanSpy whisper mode doesn't completely mute one the channels